One of the most important performance indicators for full duplex speakerphones is convergence time (i.e. the time required by the echo cancellers within the speakerphone to reach an acceptable level of cancellation). The convergence time of the speakerphone depends both on internal Line Echo Canceller (LEC) and Acoustic Echo Canceller (AEC) convergence times. In order to converge quickly and properly, a speakerphone echo canceller requires a reference signal with correct stochastic properties. At the beginning of a call (Start-up), the reference signal is usually not sufficiently stochastic (e.g. the line signal typically comprises narrow band tones such as dial tone) or speech is not present, so that echo cancellation is unable to commence immediately. In such situations the speakerphone loop may remain unstable for a noticeable period of time. This can result in feedback or “howling” of the speakerphone during start-up, especially when the speaker volume is high.
In order to prevent such feedback, it is an objective of speakerphone design to ensure that the echo cancellers (LEC and AEC) converge rapidly to the correct echo path models at start-up. Otherwise, the speaker volumes must be reduced during start-up, which may be annoying to a user.
According to one prior art approach to reducing the problem of feedback during speakerphone start-up, howling detection has been used (see ITU-T Recommendation G.168) in combination with gain control. According to this approach, the speaker volume (or loop gain) is reduced when howling is detected. A drawback of this approach is that the gain switching is often audible which may be annoying to the user.
Another prior art solution involves operating the speakerphone in a half duplex mode on start-up in order to prevent howling and echo from interfering with communication. The speakerphone remains in the half-duplex mode until the LEC adapts sufficiently to ensure echo cancellation. A drawback of this approach is that the speakerphone sometimes stays in the half-duplex mode for a long time, making communication between telephone parties difficult or impossible.
Yet another prior art solution involves forcing the speakerphone to start operation at a predetermined “acceptable” low volume level which guarantees stability in the audio loop, and then gradually increasing the volume as convergence of the echo canceller is achieved. A drawback of this approach is that the volume adjustment is often noticeable to the user.
Since the LEC models a network echo path where the first echo reflection of the near end hybrid is usually reasonably constant for each connection, and the AEC models an acoustic echo path where direct acoustic coupling or coupling through the plastic housing of the phone is always the same for a given phone, both the LEC and AEC may be loaded initially with previously captured and saved constant echo path models represented by default coefficients, and then continue to converge toward the complete echo channel models. This results in faster convergence time, and more stability as the main, strongest echo reflections will already be cancelled using the default coefficient models.
Thus, according to copending Patent Canadian Patent Application No. 2,291,428, a method is provided for improving the start-up convergence time of the LEC filter, thereby resulting in a total reduced convergence time for the speakerphone. This method is based on capturing the LEC coefficients once the LEC has converged, and saving them as the default coefficients for the next call. As a result, the echo-canceling algorithm does not have to wait for a suitable reference signal to commence convergence. At start-up, the echo canceller immediately begins canceling the line echo, based on the previously stored LEC coefficients, thereby assisting the AEC algorithm by eliminating residual line echo from the acoustic signal which the AEC algorithm is required to converge to, and initially making the speakerphone loop more stable. As indicated above, the same principal may also be applied to the AEC for direct acoustic coupling or coupling through the speakerphone housing plastic, which is always the same for a given phone. The default coefficients in this case represent the constant acoustic echo path from loudspeaker to microphone and may be reused for each new call. At start-up, the AEC immediately starts canceling the echo caused by direct acoustic coupling, while converging toward the complete acoustic echo path model that represents the combination of direct coupling and the specific room echo response.
The principle of saving default coefficients may also be applied to multiple loudspeaker-to-microphone echo paths for multiple-microphone directional systems, or even loudspeaker-to-beam echo paths for beamforming-based systems that perform echo cancellation on the output signal of a beamformer. In these cases, default coefficients can be reused from one instance of the AEC to the next in each different direction (e.g. angular sectors).
In order for such systems to work properly, the coefficients must be saved at appropriate times. If they are saved at arbitrary instants (e.g. at the end of a call), then there is a risk that the full-duplex echo cancellation algorithm will not be in a well-converged state at the instant of saving the coefficients. For example, the echo cancellation algorithm may be in the process of adapting to an echo path change related to the user moving his/her hand towards the telephone to press a button for ending the call. Saving the default coefficients in this case and reusing them at a later stage (e.g. for the next call) may result in poor echo canceller performance until it re-converges to a set of “good” coefficients.
As indicated above, the system set forth in Canadian Patent Application No. 2,291,428 tracks the degree of convergence of the full-duplex algorithm, and saves the default coefficients each time the convergence reaches a predetermined level. In one embodiment, the amount of echo actually cancelled by the algorithm is measured, and the coefficients are saved each time this amount increases by 3 dB from the previous save. One problem with this method is that if the full-duplex algorithm is subjected to narrow-band signals (e.g. in-band tones that are not detected fast enough), then it may reach excellent levels of convergence with coefficients that are very different from the useful wide-band echo-path coefficients. In such situations the system may never reach such a good level of convergence again with a wide-band signal, such that proper coefficients are never captured. This may result in annoying echo bursts for the far-end user each time these coefficients are used (for instance, at the beginning of each subsequent call). Another problem is that if the telephone is moved to a different location on a desk, where the direct echo path is more difficult to adapt to, then it may never be able to capture coefficients corresponding to its new location. It may therefore constantly reuse coefficients that do not correspond to those characterizing the real echo path, resulting in mediocre echo cancellation until the algorithm has a chance to re-converge to the real echo path.